Nate List
2006-Feb-21 11:16 UTC
[Asterisk-Users] Outbound Routing does not use Multiple Trunks
Hello, I have a TDM400 and currently have 2 of the ZAP Trunks configured on it. Zap/1-1 and Zap/2-1. I am Running Asterisk@home Version 2.4 with AMP version 1.10.010 In my Outbound Routing I have the Trunk Sequence set up so that 0 is Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk Sequence 0 is full, it does not open Trunk Sequence 1. I have found that this is true even if I have Trunk Sequence 0 set to a VoIP Line and Max channels is reached, it will not open Trunk Sequence 1. If i have Trunk Sequence 0 set to Zap/g0 then it will open the other Zap channels in order, but i need to be able to order the ZAP channels because they are charged at different rates. Have others experienced this issue? What should I be looking at to debug this? I have included the output below. SIP/700 initiated a call and Zap/1-1 answered, but when SIP/731 attempted a call, it just sat there and eventually hangs up. Thanks, Nate -- Executing Macro("SIP/700-8d41", "dialout-trunk|2|9******|") in new stack -- Executing GotoIf("SIP/700-8d41", "1?3:2)") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro("SIP/700-8d41", "user-callerid") in new stack -- Executing DBget("SIP/700-8d41", "AMPUSER=DEVICE/700/user") in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=700/user -- DBget: set variable AMPUSER to 700 -- Executing DBget("SIP/700-8d41", "AMPUSERCIDNAME=AMPUSER/700/cidname") in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=700/cidname -- DBget: set variable AMPUSERCIDNAME to 2002-ATA -- Executing GotoIf("SIP/700-8d41", "0?5") in new stack -- Executing SetCallerID("SIP/700-8d41", ""2002-ATA" <700>") in new stack -- Executing NoOp("SIP/700-8d41", "Using CallerID "2002-ATA" <700>") in new stack -- Executing Macro("SIP/700-8d41", "record-enable|700|OUT") in new stack -- Executing GotoIf("SIP/700-8d41", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/700-8d41", "recordingcheck|20060221-113809|1140539889.439") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060221-113809|1140539889.439: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/700-8d41", "No recording needed") in new stack -- Executing Macro("SIP/700-8d41", "outbound-callerid|2") in new stack -- Executing DBget("SIP/700-8d41", "USEROUTCID=AMPUSER/700/outboundcid") in new stack -- DBget: varname=USEROUTCID, family=AMPUSER, key=700/outboundcid -- DBget: set variable USEROUTCID to -- Executing GotoIf("SIP/700-8d41", "1?4") in new stack -- Goto (macro-outbound-callerid,s,4) -- Executing GotoIf("SIP/700-8d41", "1?6") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing NoOp("SIP/700-8d41", "CallerID set to "2002-ATA" <700>") in new stack -- Executing SetGroup("SIP/700-8d41", "OUT_2") in new stack -- Executing CheckGroup("SIP/700-8d41", "1") in new stack -- Executing SetVar("SIP/700-8d41", "DIAL_NUMBER=9******") in new stack -- Executing SetVar("SIP/700-8d41", "DIAL_TRUNK=2") in new stack -- Executing AGI("SIP/700-8d41", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Added prefix. New number: 16**9****** -- AGI Script fixlocalprefix completed, returning 0 -- Executing SetVar("SIP/700-8d41", "OUTNUM=16**9******") in new stack -- Executing Cut("SIP/700-8d41", "custom=OUT_2|:|1") in new stack -- Executing GotoIf("SIP/700-8d41", "0?16") in new stack -- Executing Dial("SIP/700-8d41", "ZAP/1-1/16**9******") in new stack -- Called 1-1/16**9****** -- Zap/1-1 answered SIP/700-8d41 -- Executing Macro("SIP/731-d09e", "dialout-trunk|2|3******|") in new stack -- Executing GotoIf("SIP/731-d09e", "1?3:2)") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro("SIP/731-d09e", "user-callerid") in new stack -- Executing DBget("SIP/731-d09e", "AMPUSER=DEVICE/731/user") in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=731/user -- DBget: set variable AMPUSER to 731 -- Executing DBget("SIP/731-d09e", "AMPUSERCIDNAME=AMPUSER/731/cidname") in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=731/cidname -- DBget: set variable AMPUSERCIDNAME to Nates Home -- Executing GotoIf("SIP/731-d09e", "0?5") in new stack -- Executing SetCallerID("SIP/731-d09e", ""Nates Home" <731>") in new stack -- Executing NoOp("SIP/731-d09e", "Using CallerID "Nates Home" <731>") in new stack -- Executing Macro("SIP/731-d09e", "record-enable|731|OUT") in new stack -- Executing GotoIf("SIP/731-d09e", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/731-d09e", "recordingcheck|20060221-113834|1140539914.441") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060221-113834|1140539914.441: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/731-d09e", "No recording needed") in new stack -- Executing Macro("SIP/731-d09e", "outbound-callerid|2") in new stack -- Executing DBget("SIP/731-d09e", "USEROUTCID=AMPUSER/731/outboundcid") in new stack -- DBget: varname=USEROUTCID, family=AMPUSER, key=731/outboundcid -- DBget: set variable USEROUTCID to -- Executing GotoIf("SIP/731-d09e", "1?4") in new stack -- Goto (macro-outbound-callerid,s,4) -- Executing GotoIf("SIP/731-d09e", "1?6") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing NoOp("SIP/731-d09e", "CallerID set to "Nates Home" <731>") in new stack -- Executing SetGroup("SIP/731-d09e", "OUT_2") in new stack -- Executing CheckGroup("SIP/731-d09e", "1") in new stack == Spawn extension (macro-dialout-trunk, s, 107) exited non-zero on 'SIP/731-d09e' in macro 'dialout-trunk' == Spawn extension (from-internal, 99******, 1) exited non-zero on 'SIP/731-d09e' -- Executing Macro("SIP/731-d09e", "hangupcall") in new stack -- Executing ResetCDR("SIP/731-d09e", "w") in new stack -- Executing NoCDR("SIP/731-d09e", "") in new stack -- Executing Wait("SIP/731-d09e", "5") in new stack -- Executing Hangup("SIP/731-d09e", "") in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/731-d09e' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/731-d09e' -- Hungup 'Zap/1-1' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/700-8d41' in macro 'dialout-trunk' == Spawn extension (from-internal, 99******, 1) exited non-zero on 'SIP/700-8d41' -- Executing Macro("SIP/700-8d41", "hangupcall") in new stack -- Executing ResetCDR("SIP/700-8d41", "w") in new stack -- Executing NoCDR("SIP/700-8d41", "") in new stack -- Executing Wait("SIP/700-8d41", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/700-8d41' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/700-8d41'
Mimmus
2006-Feb-21 12:01 UTC
[Asterisk-Users] Outbound Routing does not use Multiple Trunks
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Nate List > Sent: Tuesday, February 21, 2006 7:17 PM > > ... > In my Outbound Routing I have the Trunk Sequence set up so that 0 is > Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk > Sequence 0 is full, it does not open Trunk Sequence 1.Peraphs this bug in AMP: ############################################################### Max Channels Bug Remains. A bug has been reported because of a deprecated command that makes Asterisk@Home's calculation of maximum channels invalid. To fix it, goto AMP->Maintenance->Config Edit->extensions.conf->macro-dialout-trunk and comment out line s,7 so that it looks like this: ;exten => s,7,CheckGroup(${OUTMAXCHANS_${ARG1}}) Then insert a new line s,7 just below it which looks like this: exten => s,7,GotoIf($[ ${GROUP_COUNT()} > ${OUTMAXCHANS_${ARG1}} ]?108) Then click the Update button and reload Asterisk to activate the change. ############################################################### [from http://mundy.org/blog/index.php?p=112] Keep me informed if this solves your problem. Bye Mimmus
Apparently Analagous Threads
- Outbound problem sip chanel
- problem with outgoing calls Unable to create channel of type 'ZAP' (cause 34 - Circuit/channel congestion)
- problem with outgoing calls Unable to createchannel of type 'ZAP' (cause 34 - Circuit/channel congestion)
- cannot set outgoing cid
- Digium Wildcard TDM400P call pickup timing